1. Field of the Invention
The present invention relates to a voice over Internet protocol (hereinafter, referred to as “VoIP”) system, and more particularly to an improved method for providing VoIP call service.
2. Description of the Related Art
A VoIP system provides a new scheme of communication service, which performs a voice data communication over the Internet rather than a public switch telephone network (PSTN) as an existing communication network. Unlike an existing scheme, since the voice data communication over the Internet employs a packet-based network and therefore charges for using domestic/international telephone lines are not incurred, the VoIP system can inexpensively carry out the voice data communication. The VoIP system is capable of transmitting video information as well as audio information using an ITU-T standard H.323 protocol.
An H.323-based packet network, which provides VoIP service, is made up of a gatekeeper, an H.323 terminal being an endpoint and a gateway, and also includes a multipoint control unit (MCU) for conference service. The gatekeeper, which is the most important component in the H.323-based packet network, performs major functions relative to a registration of the terminal and the gateway within a zone, an address management, a call connection control, a resource management, etc. The zone is an area in which one gatekeeper manages the terminal, the gateway, the MCU, etc.
The gateway acts to transmit audio and facsimile data, incoming into the PSTN, to the Internet after a real-time compression and protocol conversion. The gateway can be divided into several types according to a built-in and use form. For example, there are a built-in type, a server type, a stand-alone type and the like. A built-in type gateway is embedded into a key telephone system or a private branch exchange (PBX) in the form of a card. A server type gateway is mounted on a platform such as a window network terminal (NT). A stand-alone type gateway is structured independently with terminals. The stand-alone type gateway is divided into a stand-alone mode and a TANDEM (trunk and ENM (ear & mouth)) mode according to an operating mode. The stand-alone type gateway of the stand-alone mode is the gateway, which is directly connected to a plurality of telephone terminals and the stand-alone type gateway of the TANDEM mode is the gateway, which supports an interworking between heterogeneous office lines. The stand-alone type gateway of the TANDEM mode is connected to the PBX and/or the key telephone system over an inner T1/E1 interface, a loop (loop start trunk) interface and a subscriber line circuit (SLC) interface.
However, in order for a caller to make a VoIP call, the caller must submit, often in DTMF, a central office access code along with the telephone number of the called party before a VoIP speech path can be set up between the caller, a local gateway and a remote gateway. After the speech path is set up, the calling party receives a second dial tone indicating that the call can go through. However, the above method is inefficient in that the calling party must wait a substantial period of time for the call to be set up. What is needed is a more efficient method of placing a VoIP call to a called party connected to a PSTN.